The invention concerns a method and a device for the selection of a sound algorithm for the processing of audio signals.
Modem high-fi equipment is provided with various sound programs which permit distribution of stereophonic audio signals to more than only two loudspeakers or to produce surround sound in some other way. Thus, for example, after decoding of the audio signals, these are split into five individual audio channels and are used through the so-called “virtualizer” for reproduction via only two loudspeakers. Special “virtualizers” are also known which convert audio signals for reproduction specifically through earphones.
One of the best known methods for this is the so-called “Dolby Pro Logic” method which, in the case of film material, is essentially used to be able to influence the localization of the sound. Thus, speakers are usually imaged on the center channel and the noises can be come exclusively from the back loudspeakers.
Furthermore, there is a whole class of methods which are used for simulation of acoustics. Frequently, applicable names of such methods are “echo”, “stadium”, “jazz”, “club”, etc. In this method, optimized for music signals, it is not desirable to take speech signals (singing) only from the center loudspeaker, or to emit a music signal only from the back loudspeakers which is possible when using the “Dolby Pro Logic” method.
In the successor of Dolby Pro Logic, which is called Dolby Pro Logic II, apart from the film mode, a mode for music is provided, which takes these differences into consideration.
A method is known for coding of speech from EP 0 481 374 B1. Here, a discrete transformation of a speech window is performed in order to obtain a discrete spectrum of coefficients. An approximate envelope of the discrete spectrum will be calculated in each of a large number of sub-bands and used for the digital coding of the defined envelope of each sub-band. Within sub-bands, each scaled coefficient is recalculated into a number of bits, with at least one of a multiple number of quantizers of different bit lengths. The quantizer used for each sub-band is determined for each speech window by calculation of the assignment of bits as a number of bits greater than or equal to zero, as a function of a power density evaluation for the sub-band and a distortion error evaluation for the speech window.
From EP 0 587 733 B1, a signal analysis system is known for filtering of an input sample value representing one or several signals. Input buffer means are provided for grouping the input samples into time-range/signal sample blocks. The input sample values are analysis-window-weighted samples. In addition, analysis means are present for producing spectral information as response to the time-range/signal sample value blocks, where the spectral information contains spectral coefficients, which used essentially in an even-numbered stack of time-range/aliasing-removal transformation, corresponds to time-range signal sample value blocks. The spectral coefficients are essentially coefficients of a modified discrete cosine transformation or coefficients or coefficients of a modified discrete sine transformation. The analysis means include forward pre-transformation means to produce modified sample value blocks and forward pre-transformation means to produce frequency range transformation coefficients.
From EP 0 664 943 B1, a coding device is known for adaptive processing of audio signals for coding, transfer, or storage and recovery, where the noise level fluctuates with the signal amplitude level. A processing device is present which responds to input signals in such a way that it emits either a first and second signal or the sum and difference of the first and second signals. The first and second signals correspond to the two matrix-coded audio signals of a four by two audio signal matrix, where the processing device also produces a control signal, which shows if the first and second signal or the sum and difference of the first and second signal is emitted.
A decoder is known from EP 0 519 055 B1, consisting of a receiving means for receiving a multiplicity of information formatted by delivery channels, deformation means for producing, in response to the receiving means, a deformatted representation depending on each delivery channel, and synthesis means for producing output signals depending on the deformatted representations. A divider means is arranged between the deformatting means and the synthesis means, which respond to the deformatting means and produce one or several intermediate signals, where at least one intermediate signal is produced by combination of the information from two or more deformatted representations. The synthesis means produce a particular output signal as response to each of the intermediate signals.
From EP 0 520 068 B1, a coder is known for coding two or more audio channels. The coder has a sub-band device for producing sub-band signals, a mixing device for creating one or several composed signals, and means for producing control information for a correspondingly composed signal. In addition, the coder has a coding device for producing coded information by allocating bits to one or several composed signals. Furthermore, a formatting device is present for combining the coded information and the control information into an output signal.
A speech coder is known from EP 0 208 712 B1. This speech coder contains a Fourier transform device for performing a discrete Fourier transformation of an incoming speech signal to produce a discrete transformation spectrum of coefficients, a standardization device for modifying the transformation spectrum to produce a scaled, flatter spectrum and to code a function through which the discrete spectrum is modified. In addition, a device is present for coding at least a part of the spectrum. The standardization device has a device (44) for defining the approximated envelope of the discrete spectrum in each of several sub-bands of coefficients and for coding the defined envelope of each sub-band of coefficients, as well as devices for scaling each spectrum coefficient relative to the defined envelope of the respective sub-band of coefficients.
However, in each of the known inventions it is a disadvantage that the selection of a sound algorithm must be adjusted manually. For example, if a television tone of an actually chosen television channel is processed through a Dolby Pro Logic II decoder and the television channel is switched several times between music stations and films or news, then upon each change one must manually switch between the individual audio sound algorithms which process the audio data, for example, between music mode and film mode.